A SIP client library using Web Worker architecture to handle SIP signaling separately from the UI thread. Supports multiple tabs sharing a single SIP session.

• ✅ Multi-tab Support: Multiple tabs can share a single SIP session
• ✅ Web Worker Architecture: SIP signaling runs in SharedWorker
• ✅ WebRTC Media: Handle audio/video calls with WebRTC
• ✅ Call Control: Hold/unhold, mute/unmute, call transfer
• ✅ DTMF Support: Send DTMF tones during calls
• ✅ Tab Coordination: Automatically select tab to handle media
• ✅ State Synchronization: Synchronize state between tabs

npm install sip-worker.js

# Build client library
npm run build

# Build worker
npm run build:worker

# Build both
npm run build:all

import { SipWorkerClient } from 'sip-worker.js';

const sipClient = new SipWorkerClient();

sipClient.register({
  uri: 'sip:username@domain.com',
  username: 'username',
  password: 'password',
  displayName: 'Display Name'
}, {
  server: 'wss://domain.com:443',
  secure: true
});

sipClient.on('sip_registered', () => {
  console.log('SIP registered successfully');
});

sipClient.on('call_incoming', (message) => {
  const callInfo = message.data;
  console.log(`Incoming call from: ${callInfo.remoteUri}`);
  
  // Accept call
  sipClient.answerCall(callInfo.id);
  
  // Or reject
  // sipClient.rejectCall(callInfo.id);
});

sipClient.on('call_progress', (message) => {
  const callInfo = message.data;
  console.log(`Call state: ${callInfo.state}`);
});

sipClient.makeCall('sip:target@domain.com');

// Mute/unmute
sipClient.muteCall(callId);
sipClient.unmuteCall(callId);

// Hold/unhold
sipClient.holdCall(callId);
sipClient.unholdCall(callId);

// DTMF
sipClient.sendDtmf(callId, '123');

// Transfer
sipClient.transferCall(callId, 'sip:target@domain.com', 'blind');

• SipCore: Handle SIP signaling with SIP.js
• TabManager: Manage tabs and select tab for media handling
• MessageBroker: Handle messaging between worker and tabs
• WorkerSessionDescriptionHandler: Custom SDP handler for worker

• SipWorkerClient: Main client API
• MediaHandler: Handle WebRTC media, getUserMedia, ICE

Tab A ←→ MessageBroker ←→ SipCore ←→ SIP Server
Tab B ←→      ↑
Tab C ←→ TabManager

Open index.html in browser to see full demo with:

• SIP registration
• Incoming/outgoing calls
• Call controls (hold, mute, transfer)
• DTMF keypad
• Multi-tab coordination

Or run dev server:

npm run dev

interface SipConfig {
  uri: string;           // SIP URI
  username?: string;     // Username
  password?: string;     // Password  
  displayName?: string;  // Display name
  registerExpires?: number; // Registration expiry (seconds)
}

interface TransportConfig {
  server: string;        // WebSocket server URL
  secure?: boolean;      // Use TLS
  iceServers?: IceServer[]; // STUN/TURN servers
}

• Chrome/Chromium 80+
• Firefox 74+
• Safari 13+
• Edge 80+

This project is licensed under a Custom License. See LICENSE file for details.

• Free for Omicx: This software is free to use for the Omicx product (https://omicx.vn)
• Other Commercial Use: For all other commercial applications, please contact luongld.it@gmail.com for licensing terms

1. Fork the repository
2. Create your feature branch (git checkout -b feature/amazing-feature)
3. Commit your changes (git commit -m 'Add some amazing feature')
4. Push to the branch (git push origin feature/amazing-feature)
5. Open a Pull Request

For support and questions:

• Email: luongld.it@gmail.com
• Issues: GitHub Issues